Packet mode speech communication

ABSTRACT

A packet mode (e.g. IP) group communication service layer is provided on top of a standard mainstream cellular radio network. Conceptually, the group communication layer comprises a pair of basic logical entities, an application bridge and a call processing server (CPS). The bridge and the CPS run group service applications which communicate with group service application(s) in a mobile station MS over the IP connections provided by the radio network. The CPS is responsible for control plane management of group communications. The bridge is responsible for the real-time distribution of VoIP packets to the user terminals according to their group memberships over valid connections programmed by the CPS.

FIELD OF THE INVENTION

[0001] The invention relates to communications systems, and especiallyto packet mode speech communication in communications systems.

BACKGROUND OF THE INVENTION

[0002] A mobile communications system refers generally to anytelecommunications system which enables wireless communication whenusers are moving within the service area of the system. A typical mobilecommunications system is a Public Land Mobile Network (PLMN). Often themobile communications network is an access network providing a user withwireless access to external networks, hosts, or services offered byspecific service providers.

[0003] Professional Mobile Radio or Private Mobile Radio (PMR) systemsare dedicated radio systems developed primarily for professional andgovernmental users, such as the police, military forces, oil plants,etc. PMR services have been offered via dedicated PMR networks builtwith dedicated PMR technologies. This market is divided between severaltechnologies—analog, digital, conventional and trunked—none of which hasa dominating role. TETRA (Terrestrial Trunked Radio) is a standarddefined by ETSI (European Telecommunications Standards Institute) fordigital PMR systems. U.S. Pat. No. 6,141,347 discloses a wirelesscommunications system which uses multicast addressing and decentralizedprocessing in group calls.

[0004] One special feature offered by the PMR systems is groupcommunication. The term “group”, as used herein, refers to any logicalgroup of three or more users intended to participate in the same groupcommunication, e.g. call. The groups are created logically, i.e. specialgroup communication information maintained on the network sideassociates specific user with a particular group communication group.This association can be readily created, modified or canceled. The sameuser may be a member in more than one group communication group.Typically, the members of the group communication group belong to thesame organization, such as the police, the fire brigade, a privatecompany, etc. Also, typically, the same organization has severalseparate groups, i.e. a set of groups.

[0005] A group call typically has a long duration (up to days) duringwhich communication takes place quite infrequently and each interactionis typically short. The total active traffic may be, for example, only15 minutes during a call. Each talk burst or speech item has an averagelength of 7 seconds in the existing PMR systems. Therefore, the radiochannels or other expensive system resources cannot be allocated all thetime, because the service becomes much too expensive. Groupcommunication with a push-to-talk feature is one of the essentialfeatures of any PMR network overcoming this problem. Generally, in groupvoice communication with a “push-to-talk, release-to-listen” feature, agroup call is based on the use of a pressel (PTT, Push-To-Talk switch)in a telephone as a switch: by pressing a PTT the user indicates hisdesire to speak, and the user equipment sends a service request to thenetwork. The network either rejects the request or allocates therequested resources on the basis of predetermined criteria, such as theavailability of resources, priority of the requesting user, etc. At thesame time, a connection is established also to all other active users inthe specific subscriber group. After the voice connection has beenestablished, the requesting user can talk and the other users listen onthe channel. When the user releases the PTT, the user equipment signalsa release message to the network, and the resources are released. Thus,the resources are reserved only for the actual speech transaction orspeech item.

[0006] There are typically various requirements for group communicationsin communications systems.

[0007] Call set up times must be relatively short, i.e. set up times inthe order of several seconds cannot be allowed. When a user initiates acall, or rather, a speech item, he/she should be able to start speakingat the initiation of the set up within few hundreds of milliseconds. Thelistening parties should hear the talk possibly within approximately asecond. This voice delay can be longer because a semi-duplex mechanismis used. These values are only examples.

[0008] Group communication requires traffic discipline: one talks andthe others listen. Therefore the radio interface is of a semi-duplextype. Only one direction is active at a time. The communications systemmust be able to control that only one member speaks at a time in agroup.

[0009] A user can belong to many groups at a same time. Therefore, acommunications system must be able to select and prioritize the groupthe user listens to if there are multiple group communications to theuser at the same time.

[0010] Not only to traditional PMR users, push-to-talk type of groupcalls are also attractive to several other types of users, too. Forexample, private persons might want to have talk groups, such as hobbygroups, sport groups, etc. Small business users might also use thepush-to-talk type of group communication feature for a more frequent jobrelated communication during a working day within the same work group,either inside the company or within some business community.

SUMMARY OF THE INVENTION

[0011] An object of the invention is to provide a new way to provide andmanage a packet mode speech communication service.

[0012] This object of the present invention is achieved by methods,systems, network units and subscriber equipment as disclosed in theattached independent claims. Various embodiments of the invention aredisclosed in the dependent claims.

[0013] According to one aspect of the invention, a mainstream cellularradio network functions as a radio access network on top of which apacket mode (e.g. IP based) group communication service is provided.Practically all new elements and functionalities required by theinvention are outside the radio access and mobile core networks. Theradio access and mobile core network can be used as such without anyneed for costly changes in the mainstream network elements. In otherwords, the group communication service can be run through standard mainstream radio access networks (such as Global System for MobileCommunications, GSM and Universal Mobile Telecommunications System,UMTS) such that the investment per end user in the infrastructure is lowenough and thereby attractive to the operators. In an embodiment of theinvention the group communication service is implemented as a Voice overIP (VoIP) data application on top of the Internet Protocol (IP) dataservice of the mobile radio network. Any user which is active in anIP-based group communication service, e.g. active in a group call, has apre-established logical connection through the radio access network tothe group communication service entities. For example, logicalconnections similar to the Packet Data Protocol (PDP) contexts used inthe GPRS service (General Packet Radio Service) may be used. The actualcommunication path, including the channel resources at the air interfacein the sending and receiving ends, needs to be opened and the resourcesto be reserved only for the duration of the talk item. Call set-upsignaling, authentication, agreement of encryption keys and negotiationof service parameters are not needed in the resource reservation phase,because the logical connections already exist, but the physicalresources are reserved and opened by using the signaling procedures.Thus, short connection set up times can be achieved.

[0014] In an embodiment of the invention all signaling relating tocontrolling a speech item is carried out as a user plane signalingembedded in a user traffic. In an embodiment of the invention, a usertraffic is in form of real time transport (RTP) packets. In anembodiment of the invention the embedded speech item signaling comprisesa leader packet sent in a beginning of a user traffic stream containinguser voice data packets, such as RTP packets, and a group communicationservice entity grants or rejects the speech item based on the leaderpacket. In an embodiment of the invention, a group communicationservice, upon granting a group communication service a speech item basedon the leader packet, opens a speech item communication to receivingmembers of a group by forwarding a user traffic stream containing saidleader packet and subsequent voice packets to the receiving members. Inan embodiment of the invention, the embedded speech item signalingcomprises a trailer packet sent at end of a user traffic streamcontaining user voice data packets, such as RTP packets, and a groupcommunication service entity ends the speech item based on the trailerpacket. In an embodiment of the invention, a group communication serviceentity forwards a trailer packet at end of a user traffic stream toreceiving group members in order end the speech item communication tothe receiving group members. Another aspect of the invention is a methodfor packet mode group voice communication in a communications system,comprising the steps

[0015] providing a group communication service entity on top of the saidcommunications system,

[0016] providing said group communication service entity with individualaddresses of group members in at least one group communication group,

[0017] sending voice packets from one of said group members to saidgroup communication service entity, each voice packet being addressed tosaid at least one group,

[0018] forwarding said voice packets individually to each receiving oneof said group members on the basis of said individual addresses.

[0019] Another aspect of the invention is a method for packet mode groupvoice communication in a communications system, comprising the steps of

[0020] providing group communication service entity with individualaddresses of group members of a group communication group,

[0021] creating an individual logical connection from each group memberto said group communication service entity by means of outbandsignaling,

[0022] starting a speech item in said group by sending a leader packetembedded in a user traffic stream from one of said group members to saidgroup communication service entity over said individual logicalconnection, each leader packet containing the identifier of therespective group member,

[0023] said group communication service entity either i) rejecting saidstarted speech item, or ii) granting the started speech item to said onegroup member and forwarding said leader packet and subsequent voicepackets in said user traffic stream individually to each receiving oneof said group members in said group on the basis of said individualaddresses.

[0024] Another aspect of the invention is a method of managing trafficstreams in a communications system having a packet mode group voicecommunication feature, comprising the steps of

[0025] providing a user-specific communications function for managingtraffic streams addressed to a user who is active in at least one groupcommunication group or in a one-to-one communication,

[0026] receiving a first voice packet stream related to a first groupcommunication group or a first one-to-one communication and addressed toa user who is active at least in said first group communication group orin said first one-to-one communication,

[0027] forwarding said first voice packet stream to said respectiveuser,

[0028] monitoring continuity of said first voice packet stream,

[0029] receiving at least one further voice packet stream related to atleast one further group or one-to-one communication,

[0030] forwarding no one of said at least one further voice packetstreams to said user if said first voice packet data stream iscontinuous,

[0031] forwarding one of said at least one further voice packet streamsto said user if said first voice traffic stream has been discontinuedfor a predetermined period of time.

[0032] Another aspect of the invention is a server system for providinga packet mode group communication service for a communications system,said server system comprising a group server provided on top of saidcommunications system, said group server further comprising

[0033] means for storing individual addresses of group members in atleast one group communication group,

[0034] means for receiving voice packets from said group members, eachreceived voice packet containing information identifying thecommunication group which the respective packet is addressed to,

[0035] means for granting a speech item to one group member per acommunication group in turn,

[0036] means for unicasting each voice packet received from said groupmember having a speech item in a group communication group separately toeach receiving member in said respective group communication group onthe basis of said individual addresses.

[0037] Another aspect of the invention is a server system for providinga packet mode group communication service for a communications system,said server system comprising a group server provided on top of saidcommunications system, said group server further comprising

[0038] means for identifying and authenticating a source of groupcommunication,

[0039] means for controlling that only one group member in a group talksat a time,

[0040] means for checking active group members in a group to which voicepackets from a currently talking group member are destined to and meansfor generating from an incoming voice packet an outgoing packet to beforwarded separately to each of said active group members, and

[0041] means for selecting from possible multiple incoming trafficstreams destined to one group member the one which is to be forwarded tosaid one group member.

[0042] Another aspect of the invention is a server system for providinga packet mode group communication service for a communications system,said server system comprising

[0043] at least one first group communication network entity providinggroup specific communications functions, said first group communicationnetwork entity further comprising

[0044] a data memory storing individual addresses of group members in atleast one group communication group,

[0045] means for receiving voice packets from said group members, eachreceived voice packet containing information identifying thecommunication group which the respective packet is addressed to,

[0046] means for granting a speech item to one group member percommunication group in turn,

[0047] means for unicasting each voice packet received from said groupmember having a speech item in a group communication group separately toeach receiving member in said respective group communication on thebasis of said individual addresses,

[0048] at least one second user communication network entity providinguser-specific communications functions for at least one user, wherebyany group related communication from a user managed by said second usernetwork entity being routed first to said second user network entity andthen forwarded to an appropriate first group network entity, and anyunicast voice packet from said at least one first group network entiybeing routed first to said second user network entity prior to sendingthe voice packet to the respective user.

[0049] Another aspect of the invention is a server system for providinga packet mode group communication service for a communications system,said server system comprising wherein

[0050] at least one group communication network entity providing groupspecific communications functions, said group network entity furthercomprising

[0051] means for controlling that only one group member in a group talksat a time,

[0052] means for checking active group members in a group to which voicepackets from a currently talking group member is destined to and forgenerating from an incoming voice packet an outgoing packet to beforwarded separately to user server having serving at least one activemember in said group,

[0053] a user communication network entity providing user-specificcommunications functions on a user plane for at least user, said usernetwork entity further comprising

[0054] means for identifying and authenticating a source of groupcommunication,

[0055] means for selecting from possible multiple incoming trafficstreams destined to one group member the one which is to be forwarded tosaid one group member.

[0056] Another aspect of the invention is a server system for providinga packet mode group communication service for a communications system,said server system comprising

[0057] at least one group communication network entity providing groupspecific communications functions in a user plane, said group networkentity further comprising

[0058] means for storing individual addresses of group members in atleast one group communication group,

[0059] means for receiving voice packets from said group members, eachreceived voice packet containing information identifying thecommunication group which the respective packet is addressed to,

[0060] means for granting a speech item to one group member percommunication group in turn,

[0061] means for unicasting each voice packet received from said groupmember having a speech item in a group communication group separately toeach receiving member in said respective group communication on thebasis of said individual addresses,

[0062] a user communication network entity providing user-specificcommunications functions on a user plane for at least one user, wherebyany group related communication from a user managed by said user networkentity being routed first to said user network entity and then forwardedto an appropriate group network entity, and any unicast voice packetfrom said at least one group network entity being routed first to saiduser network entity prior to sending the voice packet to the respectiveuser,

[0063] a group call processing entity responsible for control planemanagement of the group communications in said group network entity, and

[0064] a user call processing entity responsible for control planemanagement of the communications in said user network entity.

[0065] Another aspect of the invention is a network unit for managingspeech items in a communications system having a packet mode group voicecommunication feature, comprising

[0066] means for storing individual addresses of group members in atleast one group communication group,

[0067] means for receiving voice packets from said group members, eachreceived voice packet containing information identifying thecommunication group which the respective packet is addressed to,

[0068] means for granting a speech item to one group member per acommunication group in turn,

[0069] means for unicasting each voice packet received from said groupmember having a speech item in a group communication group separately toeach receiving member in said respective group communication group onthe basis of said individual addresses.

[0070] Another aspect of the invention is a network unit for managingtraffic streams addressed to a user who is active in at least one groupcommunication group or in one-to-one communication, comprising

[0071] means for selecting for unicast to a user a first voice packetstream related to a first group or one-to-one communication addressed tosaid user,

[0072] means for monitoring continuity of said selected first voicepacket stream,

[0073] means for discarding any other received voice packet streamrelated to at least one further group or one-to-one communication, ifsaid currently selected voice packet stream is continuous, and means forselecting and unicasting another received voice packet stream to saiduser if said initially selected and unicasted first voice traffic streamhas been discontinued for a predetermined period of time.

[0074] Another aspect of the invention is a method for establishing aone-to-one voice communication in a communications system, comprisingthe steps of

[0075] providing a communication server on top of the said mobilecommunications system,

[0076] creating an individual logical connection between saidcommunication server and each user having an active communicationservice in said communication server,

[0077] starting a communication by sending a leader packet embedded in atraffic stream from a user to said communication server over respectivesaid individual logical connection, each leader packet containingidentifier of said sending user and a receiving user,

[0078] said communication server either i) rejects said started speechitem, or ii) grants the started speech item to said sending user andforwards said leader packet and subsequent voice packets of said usertraffic stream to said receiving user on the basis of said receivedidentifier of said receiving user.

[0079] Another aspect of the invention is a subscriber equipment forcommunications system having a packet mode group voice communicationservice, said subscriber equipment comprising

[0080] mechanisms for packet data communication over a communicationssystem,

[0081] a group communication application on top of said mechanisms,

[0082] said application having first means for establishing a logicalpacket connection to a group communication server,

[0083] said application having second means for sending and receivingvoice packets to and from said group communications server.

[0084] Another aspect of the invention is a subscriber equipment forcommunications system having a packet mode group voice communicationservice, said subscriber equipment comprising

[0085] a push-to-talk means,

[0086] means, responsive to activation of said push-to-talk means by auser, for sending a leader packet followed by voice packets in a usertraffic stream to said group communication service and thereby startinga speech item.

[0087] Another aspect of the invention is a method for providing apacket mode group communication service for a communications system,comprising

[0088] storing individual addresses of group members in at least onegroup communication group,

[0089] managing said group communication groups using a control planesignalling,

[0090] group member requests speech item using user-plane signallingembedded in a user traffic stream,

[0091] granting a speech item to one group member per a communicationgroup in turn based on said embedded user plane signalling,

[0092] receiving voice packets from a group member having a speech itemin a group communication group, each received voice packet containinginformation identifying the communication group which the respectivepacket is addressed to,

[0093] unicasting said embedded user-plane signalling and each voicepacket received from a group member having a speech item separately toeach receiving member in said respective group communication group onthe basis of said individual addresses.

[0094] Another aspect of the invention is a server system for providinga packet mode group communication service for a communications system,comprising

[0095] means for storing individual addresses of group members in atleast one group communication group,

[0096] means for managing said group communication groups using acontrol plane signalling,

[0097] means for granting a speech item to one group member per acommunication group in turn based on speech item requests speech sent bysaid group members using user-plane signalling embedded in a usertraffic stream,

[0098] means for receiving voice packets from a group member having aspeech item in a group communication group, each received voice packetcontaining information identifying the communication group which therespective packet is addressed to,

[0099] means for unicasting said embedded user-plane signalling and eachvoice packet received from a group member having a speech itemseparately to each receiving member in said respective groupcommunication group on the basis of said individual addresses.

BRIEF DESCRIPTION OF THE DRAWINGS

[0100] In the following, the invention will be described in greaterdetail by means of preferred embodiments with reference to theaccompanying drawings, in which

[0101]FIGS. 1, 2 and 3 illustrate the basic architecture of theinvention;

[0102]FIGS. 4 and 5 are signaling diagrams illustrating the allocationof uplink and downlink bearers, respectively, in the radio interface ofa mobile network;

[0103]FIG. 6 shows an overview of a group management concept;

[0104]FIG. 7 is a signaling diagram illustrating user log-on to PoCservices;

[0105]FIG. 8 is a signaling diagram illustrating signaling relating tothe management of a group speech item;

[0106]FIG. 9 is a flow diagram illustrating the management of a PoCgroup speech item by talkspurt timers;

[0107]FIG. 10 is a flow diagram illustrating the downstream suppressionby an upstream timer;

[0108]FIG. 11 is a block diagram illustrating user-plane groupcommunication with two bridges involved;

[0109]FIG. 12 is a diagram illustrating the multi-unicast concept;

[0110]FIG. 13 is a block diagram illustrating the scanning filteringprocess, and

[0111]FIG. 14 illustrates an implementation of the scanning filteringprocess,

[0112]FIG. 15 is a signaling diagram illustrating signaling andcommunication relating to the setup of one-to-one communication.

PREFERRED EMBODIMENTS OF THE INVENTION

[0113] The present invention is applicable to any digital communicationssystems which can be used as an access network allowing packet modecommunication between end users and an overlaying packet mode groupcommunication service. The invention is especially preferably used inmobile communications systems based on a GPRS-type packet radio. In thefollowing, the preferred embodiments of the invention will be describedby means of a GPRS service and the UMTS or GSM system without limitingthe invention to this particular packet radio system. The IP voicecommunication method used in the preferred embodiments of the inventionis the Voice over IP (VoIP), but the invention is not limited to thisparticular method.

[0114]FIG. 1 illustrates the basic architecture of the preferredembodiment of the invention. In the illustrated embodiment, a mobileRadio Access Network (RAN) which provides the IP packet data service isbased on a GPRS architecture utilizing a 2G radio access technology,such as a GSM Base Station Subsystem BSS with Base Transceiver StationsBTS and Base Station Controllers BSC. The GSM radio access may beconventional or based on the GSM Enhanced Data rates for GSM Evolution(EDGE) technique. In the latter case, radio access may be referred to asGERAN which is an all-IP GSM radio access network. Alternatively, a 3Gradio access network UTRAN (such as UMTS) may be used. An all-IP corenetwork can be used both in GERAN and UTRAN. The architecture of themobile network is not essential to the invention, but the GPRSinfrastructure and operation will be briefly discussed in order to makeit easier to comprehend the invention. The GPRS infrastructure comprisessupport nodes, such as a Gateway GPRS Support Node (GGSN) and a ServingGPRS Support Node (SGSN). The main functions of the SGSN are to detectnew GPRS mobile stations in its service area, handle the process ofregistering new Mobile Stations MS (also called User Equipment, UE)along with the GPRS registers, send/receive data packets to/from the MS,and keep a record of the location of the MSs inside of its service area.The subscription information is stored in a GSM/GPRS register (HLR, HomeLocation Register or in 3G all-IP networks HSS, Home Subscriber Server).The main functions of the GGSN nodes involve interaction with externaldata networks. The GGSN may also be connected directly to a privatecorporate network or a host. The GGSN includes PDP addresses and routinginformation, i.e. SGSN addresses for active GPRS subscribers. The GGSNupdates the location directory using routing information supplied by theSGSNs. The GGSN uses the routing information for tunneling the ProtocolData Units (PDUs) from external networks to the current location of theMS, i.e. to the serving SGSN, in accordance with the GPRS TunnelingProtocol (GTP). Tunneling means that the data packet is encapsulatedinto another data packet during transfer from one end of the tunnel toanother. The GGSN also decapsulates data packets received from MSs andforwards them to the appropriate data network. In order to send andreceive GPRS data, the MS activates the packet data address that itwants to use, by requesting a PDP activation procedure. This operationmakes the MS known in the corresponding GGSN, and interworking withexternal data networks can commence. More particularly, one or more PDPcontexts are created and stored in the MS and the GGSN and the SGSN. ThePDP context defines different data transmission parameters, such as PDPtype (e.g. X.25 or IP), PDP address (e.g. IP address) and Quality ofService (QoS).

[0115] In FIG. 1, a Push-to-talk Over Cellular (PoC) layer is providedon top of the mobile network in order to provide group communicationservices to the Mobile Stations (MS) through the mobile network.Conceptually, the PoC layer comprises a pair of basic logical entities,a PoC bridge 10 and a PoC Call Processing Server (CPS) 11. The bridge 10and the CPS 11 are connected to the GGSN, typically over an IP network.The bridge 10 and the CPS server 11 run PMR applications whichcommunicate with the PMR application(s) in the mobile station MS overthe IP connections provided by the IP mobile RAN. This communicationincludes both signaling packets and voice (group and one-to-one)communication packets.

[0116] The CPS 11 is responsible for control-plane management of the PMRcommunications. Its important role may require various functionalitieswhich in an embodiment of the invention are implemented in the followingmodules: “PMR server”—the application that handles the sessions forgroup memberships which are signaled with an appropriate session controlprotocol, such as Session Initiation Protocol (SIP), established for thePoC communications, and manages the users profiles (call rights, groupactive membership, scanning settings, etc.); SIP Proxy/LocationServer—providing user location and routing functionalities of SIPsignaling; SIP Registrar—for user registration/authentication; and MediaGateway Controller—controlling the network entities (PoC bridges)involved in the IP layer data distribution according to the group & userspecific information (membership, rights, scanning settings, etc.).However, in this description, the common term CPS refers to all possiblefunctionalities of the CPS.

[0117] However, since the PMR management requirements can be dividedinto group and user specific ones, two kinds of CPS servers are definedin one embodiment of the invention, as illustrated in FIG. 2. The SIPsessions for group communications are handled by a Group Control PlaneFunction (G-CPF) (G-CPF) 23 (e.g. in a server). When a user attaches toa group, the G-CPF 23 takes care of the relative SIP invitationtransaction and performs the proper mapping settings between the user'srecipient and the network entities responsible for the relative trafficdistribution. The User—Control Plane Function (U-CPF) 22 (e.g. a controlplane proxy server) is basically the control plane interface between theIP network and the user. By this network entity the users log on to thesystem and negotiate their operational settings (scanning settings,selected group etc.). It handles the user's profile and manages hisone-to-one calls. It should be appreciated that this is just a logicalseparation, and both kinds of CPS can be situated in the same computer.Separating G-CPF and U-CPF enables users to join PoC groups handled byG-CPF in different intranets or in mobile networks of differentoperators and IP domain. Division also brings scalability by allowing inpractice infinite number of groups or users in the system.

[0118] Referring again to FIG. 1, the bridge 10 is responsible for thereal-time distribution of VoIP packets to the users' terminals accordingto their group memberships, their scanning settings and eventualpre-emption or emergency cases. Each bridge forwards traffic onlybetween valid connections programmed by the CPS. The bridge 10 mayperform one or more of the following functionalities:

[0119] Input checking: to identify and authenticate the traffic source(optionally the mnemonics in the leader RTP packet, which will bediscussed below, have to be processed here). Input checking may alsoinclude actions to perform and support security procedures.

[0120] Input filtering: to manage that only one talker talks in a groupat a time (i.e. grants a speech item), and optionally to give priorityto higher priority voice items.

[0121] Multiplication: after the filtering process, the bridge 10 has tocheck the active members of the group to which the traffic is destinedand generate from the incoming packet a “downlink” packet for eachactive member.

[0122] Scanning filtering: to select from the multiple incoming trafficstreams destined to the same user the one which has to be forwarded tohis recipient according to the user's scanning settings.

[0123] Again, since input filtering and multiplication are groupspecific processes, while input checking and scanning filtering are userspecific, the following two kinds of application bridges have beendefined in one embodiment of the invention, as illustrated in FIG. 2.

[0124] Firstly, a Group—User Plane Function (G-UPF) G-UPF 21 (e.g. in aserver) is a network entity to which group members' audio packets aresent (through their U-UPF) and where the input filtering andmultiplication processes are performed. To each new group the G-CPF 23assigns a single G-UPF 21 according to load balancing criteria whichdistributes the traffic as evenly as possible between the G-UPFs.

[0125] The User—User Plane Function (U-UPF) U-UPF20 (e.g. in a server)performs the input checking and scanning processes for the individualsubscribers which have been assigned to it by the U-CPF 22. For securitypurposes the U-UPF 20 may have security associations for each mobileterminal it handles. The U-UPF 20 hides the network complexity from themobile terminals, so the user has just to send all his user planetraffic to this unit that afterwards forwards it according to themapping settings of the proper U-CPF 22. In this way there is no need toestablish secure channels between each user and all the IP networkentities which have just to trust the U-UPF 20 from which they receivepackets.

[0126] As for the Control Plane elements, this logical splitting doesnot necessarily require a physical separation between the G-UPF and theU-UPF implementations, and thus they may be located in the samecomputer.

[0127] The U-CPF 22 and the G-CPF 23, which are responsible for managingthe sessions of the users and the groups, respectively, require specificcontrol plane signaling. ETSI 3GPP (European TelecommunicationsStandards Institute, 3rd Generation Partnership Project) specificationsinclude IP based voice communications in a so called all-IP network.Such an all-IP network enables also voice communication in IP network(voice over IP, VoIP). For VoIP, call control signaling is specified,such as the Session Initiation Protocol (SIP), which is defined in theRFC2543. Therefore, in the preferred embodiment, the SIP has been chosento support and manage the PoC call sessions. However, some other IPsession protocol may be used instead. Further, in the preferredembodiment of the invention, Megaco (defined in RFC3015) is used by theG-CPFs 23 and the U-CPF 22 to control the G-UPFs 21 and U-UPFs 20involved in traffic distribution of the IP layer. However, some othercorresponding protocol for controlling the switching of the user planeelements may be used instead. Still further, RTP (Real-time TransportProtocol, defined in RFC1889) has been chosen to handle the transfer,and QoS mechanisms are needed to handle the voice packet (VoIP)delivery.

[0128] Megaco defines a general framework for physically decomposedmultimedia gateway. Its connection model is based on two mainabstractions which are Termination and Context. The former is a logicalentity in the MGW (i.e. PoC Bridge) that sources and/or sinks one ormore streams, while the latter is an association between a collection ofTerminations that describes the topology (who hears/sees whom) and themedia mixing and/or switching parameters if more than two Terminationsare involved in the same association. Priority values can be used by theMGC (i.e. PoC CPS) in order to provide the MGW with information about acertain precedence handling for a context, and an indicator for anemergency call is also provided to allow a preference handling. Theprotocol provides commands for manipulating the logical entities of itsconnection model, contexts and terminations, and it is here assumed thatit provides the flexibility and the functionalities required by the PMRCPS 11 (the G-CPF 23 and the U-CPF 22) to program the proper trafficpaths and filtering/scanning processes in the PoC Bridge 10 (the G-UPF21 and the U-UPF 20).

[0129] The SIP protocol defines signaling messages for call control,user location and registration, and these have been used in thepreferred embodiment of the PoC solution to handle the specific PMRcommunications and the relative participating users (establishment,joining and tear down of a call session, user's log on to PoC services,user's profile negotiation, etc).

[0130] For each PoC communication, a SIP session is established andmanaged by the CPS handling it (G-CPF 23 and U-CPF 22 for group andone-to-one communications respectively). When a user wants to become anactive member of a group, he has to join the corresponding session. Forone-to-one calls, the PoC U-CPFs maintain one session betweenparticipating U-CPFs for each one-to-one call.

[0131] All the user's outgoing and incoming traffic has to go throughthe U-UPF 20 that has been assigned to the user. In particular, in theuplink the user's traffic is checked by his U-UPF 20 and forwarded tothe G-UPF 21 handling the group to which the traffic is destined or, incase of one-to-one communication, to the U-UPF 20 handling the calledparty.

[0132] In the downlink, the traffic is then distributed to thedestination users' U-UPFs 20 (by packet multiplication in the G-UPF 21in case of group communication, packets are multiplied and forwarded toeach U-UPF which is serving active members in the group). In the U-UPF,the users' scanning processes are performed and traffic is multipliedand forwarded to each user that listens to the group according to hiscurrent scanning settings.

[0133] This PoC solution is access independent, which means that it canrun on top of GSM, WCDMA, WLAN or equivalent technologies as long asthese are able to support the always-on VoIP bearers. The IP layer'saudio distribution uses standard VoIP mechanisms (such as the RTP),while specific Internet protocols or interfaces will be used to connectsupplementary network entities, such as Subscriber and Group ManagementFunction (SGMF) 25, a Domain Name Server (DNS) 24, WWW/WAP (World WideWeb/Wireless Application Protocol) and security management servers. Eachnetwork entity is obviously associated with at least one IP address bywhich the IP packets are transferred and routed, but the role of thenetwork elements have also to be defined from the SIP's point of view.Each MS is a SIP User Agent (UA), and thus each one has a SIP address(URL) which normally is “username@hostname” where the hostname can be,but not necessarily is associated with the U-CPF 22 in which the MSshave to register. This U-CPF 22 should act as a Registrar, Location andProxy SIP server in order to allow the reachability of the MSs under hiscontrol and to support the SIP signaling routing. The G-UPFs 21 andU-UPFs 20, which are exclusively involved in the audio datadistribution, do not have a role in the actual SIP mechanisms and thecore network is simply seen as a single IP network link. At the SIPsignaling level, URLs are used for user and group identification. TheURLs can be sip: URLs as defined in the RFC2543, tel: URLs representingtelephone numbers as defined in the RFC 2806, or any other URL formats.The REGISTER method is used with a sip: URL, that is, SIP URL is theuser main identity in PoC system. Dialing of users with a privatenumbering plan number (only) is possible using the tel: URL in the To:header field (sip: URL must have the host portion present at all times).A secondary identity can be used for example for addressing the b-partyfor one-to-one calls if the b-party is from the same Virtual PrivateNetwork (VPN). Groups are always addressed with sip: URLs, where thegroup name is used in place of the user name, and the host managing thegroup (exact G-CPF, if known) in the host portion. The addressing on theuser plane will be explained in more detail below.

[0134] Additionally, an SGMF 25 is preferably provided in PoC system formanagement and information query/updating purposes. Via SGMF 25,operator or a normal user having management rights can create, deleteand modify users and groups in PoC system. Also access rights related tousers and groups can be created and modified. The information itself canbe contained in a database, such as Structured Query Language (SQL)database or in a directory, such as Lightweight Directory AccessProtocol (LDAP, defined in RFC2251) directory. These data repositoriescan be stand-alone or co-located with SGMF 25. This database ordirectory is the main data repository in PoC system. Normal users havingmanagement rights can access SGMF using a WWW/WAP interface. Animportant function of SGMF 25 is also processing requests coming fromU-CPF 22 and G-CPF 23 and making database or directory fetches andupdates according to the requests.

[0135] SOAP (Simple Object Access Protocol, defined by the World WideWeb Consortium W3C), or a similar protocol can be used in the interfacebetween U-CPF 22 and SGMF 25 as well as in the interface between G-CPF23 and SGMF 25.

[0136] The user equipment, or Mobile Station MS, has a PoC applicationon a user layer on top of the standard protocol stack used in thespecific mobile communications system. The SIP and RTP protocols employthe underlying Transmission Control Protocol (TCP), User DatagramProtocol (UDP) and IP protocols which further employ the physical layerresources, such as the radio resources. Additionally, a WAP stack may beemployed to access the WAP pages on SGMF 25 or on some another server.

[0137] In FIG. 3, one possible general PoC architecture is presented.The IP backbone 29 may be, for example, an IP mobile backbone, a LAN, aPoC intranet, or two or more separate intranets, etc.

[0138] PoC mobile MS, when the PoC mode is selected by the user sets uptwo GPRS contexts: a) one to the PoC CPS 11 to be used with TCP/IP forcontrol plane signalling (group management, registration etc.), b) onefor voice to/from the PoC bridge 10 using RTP,UDP, conversational IPquality class or similar, and sufficient header compression over theradio path. If a mobile or the mobile network do not support twosimultaneous contexts, the mobile must clear down the RTP connection forthe duration of the SIP signaling transaction. The PoC mobile MS mustalways maintain the contexts to the bridge 10 when the PoC mode is on.The SIP content is also preferably on all the time, but if this causesproblems to network capacity or to the accessibility of other servicesthan PoC, the SIP context can be set up also for the duration ofsignaling transactions. Notice: in this case the cellular network mustsupport the network initiated context set up. The SIP sessions aresignaled in power on or in PMR mode activation. The SIP sessions arealways on and thus no SIP signaling is needed for PMR voice items. Allvoice is transmitted after PTT activation via the existing contexts.This mechanism enables fast call set up.

[0139] An example of the allocation of the uplink bearer at the radiointerface of the mobile RAN is illustrated in FIG. 4. The user pushesthe PTT and the MS sends a speech item request to the mobile RAN. The MSwill ask for a dedicated radio bearer for the duration of whole speechitem. The mobile RAN grants the uplink bearer (e.g. a dedicated packetdata channel and the physical time slot). When the mobile RANacknowledges allocation of the uplink bearer, the mobile starts sendingdata through it. The first packet sent is an RTP message containing thetalking party identifier followed by voice stream packets (VoIP RTPpackets). The leader RTP packet and the VoIP RTP packets are routed tothe PoC bridge 10 on the basis of the active GPRS context.

[0140] The PoC bridge 10 multiplies the packets and sends them to theother members of the group. An example of the allocation of the downlinkbearer in the radio interface of the mobile network is illustrated inFIG. 5. The downlink bearer is allocated by the SGSN when it detects anIP packet going via an existing context to a mobile station MS. Firstly,the SGSN pages the MS if it is in a STANDBY state. After receiving anacknowledgement from the MS, the SGSN requests that the RAN (e.g. theGSM BSS) allocates a dedicated radio bearer, and after the allocationthe SGSN starts sending packets (e.g. in LLC frames) to the RAN. The RANsends the packets (e.g. in radio blocks) to the MS.

[0141] The uplink voice bearer is released by the MS when the user stopspushing the PTT switch. The network will release the uplink bearer whenthe maximum speech item length (e.g. 20 to 30 sec) is exceeded. In thedownlink direction the radio network may release the bearer when no IPmessages associated with the bearer have been received for apredetermined period of time (so called idle timeout).

[0142] The call set up delay experienced by the caller after pressingthe PTT switch may be shortened by the mobile station MS giving anaudible indication to the user to start speaking. After the audibletone, the user can start speaking and the VoIP message starts. This isthe time the caller experiences as the set up delay. There are severalpoints at which the permission to speak can be given. For group calls,one suitable point is after the uplink radio bearer has been allocatedand after the first RTP message (so called leader packet, non-voice) hasbeen sent to the RAN.

[0143] In one-to-one calls, the indication to start speaking can furtherbe received from the called party. Notice that when the first RTP packetis sent to uplink, the downlink status is not known at that point. Incase of call failure because of a missing B party or missing radiobearers in the downlink direction or a failure of a call authorizationcheck, the user gets an indication of a call failure. The indication tospeak could be alternatively given after the bridge 10 gives anacknowledgement of, for example, having processed the first RTP packetor, in the one-to-one calls, after the B party has acknowledged theleader packet. Still alternatively, the MS could have a timer value setby the CPS from sending the lead packet to giving the audible indicationto the user.

[0144] Group Communication

[0145] Groups (also called talkgroups) provide the users with an easyand immediate multipoint way for voice communication. Each user can beallowed access to one or more groups. A typical case is that a mobileuser is allowed access to all groups in his Virtual Private Network(VPN). The user can be actively attached to a subset of the availablegroups.

[0146] In the basic mode, the mobile user selects one group forcommunication. He will then hear all traffic in that group (unless he isengaged in an individual call) and can also talk in the group. The usercan easily switch to another group.

[0147] The user can also operate in multiple groups virtually at thesame time, by using a method called scanning. The user selects multiplegroups and assigns these with priorities. He then hears traffic from onegroup at the time, but traffic from a more important group willinterrupt other traffic. One of the groups remains the selected group,and any speech transmission by the user is made to the selected group.The user can switch scanning on and off. The list of scanned group withpriorities can be edited by the user. Group selection and changing ofother settings can also be performed by someone else than the userhimself.

[0148] The user interface for receiving and talking in groups, changingthe selected group and activating scanning is simple and fast. Othertasks, such as defining the scanning list are used less often.

[0149] PMR-Style One-to-One Communication

[0150] As an option, the architecture according to the invention can beused to enable the users to make direct one-to-one calls to other userswithin their defined access rights (default: within their VPN). A directone-to-one call resembles the use of an intercom rather than the use ofa normal telephone. Such calls are well suited to many PMR users: tasks,commands and advice can be given and received with minimal attention tooperating the mobile station. Activities can be coordinated with goodtiming accuracy without having to keep a call on during long periods.Basically, an one-to-one call is only a special case of groupcommunications, and the same principles can be used.

[0151] Management Plane Operation

[0152] In the following, the preferred embodiments and different aspectsof the invention are discussed on the management plane, control planeand the user plane of PoC.

[0153] User and Group Management

[0154] With MS equipment, users may be able to browse the possiblegroups and subscribe to them. They may also be able to leave the groups.For more professional use, forced joining and removal to/from groups isneeded. It is desirable that the group management is produced via aWEB/WAP browser based service.

[0155] First of all, users must be created in PoC system. This is doneby accessing SGMF 25 using a WAP/WWW interface. All user and groupmanagement operations can be performed by a management user, who canaccess SGMF using a MS or can be directly connected to SGMF.

[0156] Secondly, groups need to be created before they can be used forcommunication. Creating groups and defining their access rights belongsto what is called group management. Many user groups or end userorganizations are expected to outsource their group management, but somewill prefer to have access to creating groups and defining group membersand access rights. On the other hand, not all users need to create newgroups (e.g. ordinary workers using PoC). Therefore, it is better tohave a separate concept of management user in the PoC system. In thepreferred embodiment of the invention the users can have a remote accessto a Subscriber and Group Management Function (SGMF) 25 provided by theoperator and shown in FIGS. 2 and 3.SGMF may provide a group managinguser interface using WAP/WWW forms. However, other types of userinterfaces are also possible.

[0157] An overview of the group management concept is shown in FIG. 6.Group management is used by management users 61 to create groups for theuse of users 62. The users 62 can be actively engaged in a group (anactive group session is established) or they can have groups bookmarkedin the group list 63 of their MSs for easy use later. Furthermore, theusers 62 can have been allowed access to yet other groups. A user 62 canactivate a session in such groups, e.g. by typing the URL of the group(such as ‘football@publicgroups.operator.fi’) or clicking a link on aweb or WAP page. The management users 61 can be either 1) normal userscreating or modifying groups for personal or business use, 2) officepersonnel creating or modifying groups for company use, 3) dispatcherscreating or modifying groups for their PMR fleets, or 4) operator orservice provider personnel creating or modifying groups for theircustomers' use.

[0158] First of all, SGMF 25 must hold information on authorizedmanagement users and what they are allowed to do. The information mayinclude settings like: 1) which operation the management user ispermitted to use (e.g. create, add/remove access rights, sendnotifications); 2) which groups he is allowed to manage (e.g. ownprivate groups, any groups of company-k, any public groups ofprovider-x); and 3) which users he is allowed to include (e.g. any, anyusers of company-k, a list of persons) in the groups.

[0159] Then let us consider an example case wherein an authorizedmanagement user 61 creates a group. In an embodiment of the invention,the group data created at this point may include: 1) the home CPS 23 ofthe group; 2) the URL of the group (dependent on the home CPS 23); and3) the initial access rights settings for the group (can be changedlater). The group creation/management application may now perform e.g.the following actions: 1) update the DNS server 24 of the URL ifnecessary (typically there should be no need if existing domain namesare used); 2) update the CPS 23 with the group name; 3) and update thePoC database or directory (PoC main information repository) 65.

[0160] The management user 61 may at this point also want to send anotification of the new group to potential group members. For instancewe can see the following typical cases: 1) the management user 61 is aprivate person who has created a group for five persons he knows, accessto the group has been restricted to these five persons, and the userwants notification to be sent to these five persons; 2) the managementuser is a service provider who has created a group for hobbyists, accessto the group has been set open to all, and the notification is sent to alist of users who according to marketing research are likely to beinterested.

[0161] The notification of a new group is, for example, a special formof SMS message (e.g. ring tones, logos), or a SIP instant message. TheMS may react to this message by e.g. 1) displaying to the user that anew group is available to this user; 2) giving the MS user a choice ofjoining immediately (starting an active session; normal or sticky) orbookmarking for later use, or rejecting (a reject message will be sentto SGMF which may display it to the management user). The rejectionindicates to the application that the user does not accept the group,but this does not necessarily have to result in modifications to accessrights data.

[0162] As noted above, new groups will be added by the SGMF to therelevant G-CPF 23. Likewise the SGMF can also delete groups. The G-CPF23 is not directly involved in creating groups otherwise. Afternotifications have been sent to users, the users who wish to join thegroup immediately appear to the G-CPF 23 as users establishing a SIPsession to a group. Now, G-CPF 23 inquires group access rights from theSGMF 25 which in turn makes an inquiry to PoC database or directory (PoCmain information repository) 65.

[0163] The removal of a user's group access rights affects only the PoCdatabase or directory 65. Any ongoing sessions are therefore notaffected, and the change becomes effective at the next session set-up.If a user has to be removed from a group, a separate facility for thatmay be implemented to the G-CPF 23. The deletion of a group is indicatedby the SGMF 25 to the appropriate G-CPF 23. The G-CPF 23 will then endall active sessions and remove any stored information on the group. TheSGMF 25 also takes care of removing information in the PoC database ordirectory 65.

[0164] Group access rights are checked by the CPS at the time when agroup session for user equipment is started. Additional checks can bemade at other times if deemed necessary to maintain security. In thepreferred embodiment of the invention, the group access rights are heldin the database or directory 65 which is then inquired by an appropriateserver. The typical inquiry takes the form “is user-x allowed to accessgroup-y?”.

[0165] The access rights definition is preferably flexible and possibleboth on the level of individual users/group and on lists ofusers/groups. For instance, one should preferably be able to define: 1)user-x allowed to access group-w; 2) user-x, user-y, user-z allowed toaccess group-w; 3) user-x allowed to access all groups of company-k; 4)all users of company-k allowed to access group-z; 5) all users ofcompany-k allowed to access all groups of company-k; 6) all usersallowed to access group-p; 7) etc.

[0166] Therefore, the access management preferably uses a hierarchicalstructure for both users and groups. This means that users can belong touser groups and groups can belong to group groups, even on multiplelevels. It would also be even more flexible if a single user couldbelong to multiple (parallel) user groups. Group access can be given toa specific user or to an entire user group. Access given to a user groupadmits all users in that user group. A user can be given access to aspecific group or to a group group. Access to a group group admits intoall groups in that group group.

[0167] Control Plane Operation

[0168] User Log on to PoC Services

[0169] Before the user can start to use PoC services he has to registerhimself to his U-CPF 22 whose actual IP address has to be determined byDNS services. In the preferred embodiment of the invention the userfirst makes a DNS query containing the host part of his SIP address. TheDNS 24 returns the IP address of the U-CPF 22 corresponding to the hostpart.

[0170] Referring to an example shown in FIG. 7, once the MS knows the IPaddress of the U-CPF 22 it sends a SIP registration message to the U-CPF22. When the U-CPF 22 receives the registration message from the user'sMS it contacts SGMF 25 for checking rights of the user and obtainingother information. After this, U-CPF 22 contacts U-UPF 20 of the userwhere his input checking and scanning filtering process has to beperformed and where the user has to send his user plane traffic. Theuser is then added to the U-UPF by an Add message, and the U-UPFinitializes the user's scanning process and sends an acknowledgement.Optionally, before contacting the U-UPF 20, the U-CPF 22 may exchangeuser information with the Home Location Register (HLR) or 3G HomeSubscriber Server (HSS) of the user, authenticate the user and create auser profile.

[0171] During the logon the user gets the IP address of his U-UPF 20,and possibly a list of user's sticky groups (explained later).

[0172] The registration message normally includes the identificationinformation of the user, but the message can also include other relevantindications. It can be re-sent by the user in order to make a new logonand to request particular information from his U-CPF 22.

[0173] In order to avoid the log-on of different users to the system bythe same terminal, which would require more than one scanning processesfor the same IP recipient, a specific checking mechanism may beperformed by the U-CPF 22.

[0174] In case the user wants to update his scanning settings or setON/OFF his scanning process, then he can send new specific SIPregistration messages to his U-CPF 22. If the user wants to select acertain group (scanning being set off), it is done by sending a SIP INFOmessage to user's U-CPF 22. In case the user has sticky groups(permanent groups), they are activated at logon: the U-CPF 22 performsthe consequent operations required, such as SIP session invitation,mapping settings in the G-UPFs 21 and the U-UPF 20, and finally providesthe resulting information (for example the list of the sticky sessionswhich the user has implicitly joined) to the user in the logonacknowledgement message.

[0175] Active Group Sessions

[0176] A user communicates (listens and talks) in groups for which hehas an active session. Sessions are set up and ended by SIP signaling.The session setup can be initiated both by the user or by an authorizedthird party (such as a dispatcher or an application). Sessionestablishment by a third party is mainly relevant only in PMR use. Manyusers, especially in the non-PMR market are likely to dislike sessionestablishment by a third party and may like to be able to prevent this.The sessions may also be forcibly ended by the G-CPF 23, e.g. in case ofgroup deletion.

[0177] The primary and effective data on the active group sessions isalways held by the server(s). Thus, if the user equipment (e.g. the MS)has lost data on active sticky group sessions, it can request allnecessary group information from U-CPF by performing a new logon.

[0178] In many applications the user may continue using the same groupsafter a power-off period. For this purpose, sticky sessions areprovided. When the user is logged off, the information related to user'ssticky sessions is saved in PoC database or directory (PoC maininformation repository), and the sessions are re-established at poweron. In other words, sticky group is talk group that is automaticallyactivated after a new logon.

[0179] For activating a group session, the MS needs to know the URL ofthe group. From the user's point of view, he may (user decision, dependson what options have been implemented) select the group by

[0180] 1) typing the full URL of the group (e.g.sector2@hkl.grpcps.operator.fi, football@publicgc.operator.fi).

[0181] 2) selecting from groups stored in the MS in a group bookmarklist.

[0182] 3) using a WAP/WWW application to browse available groups.

[0183] All these methods are complementary and can be compared tocorresponding methods in web browsing: typing the URL, selecting fromthe bookmark list, clicking on a link on a web page. The outcome in allthree cases is that the MS knows the URL of the required group and canstart SIP signaling.

[0184] Setting up a session by the user may (if this facility isimplemented and the user decides to use it) be based on an URL of thegroup given by the user. This allows any user to try access to anygroup; access rights checking will then be performed by the server(s).Another method for occasional access to groups would be using a web/WAPbrowser to browse for interesting and/or useful groups. Both of thesemethods are very suitable to occasional and temporary access to groups.

[0185] However, if the user needs frequent access to some groups withouthaving to keep the session open all the time, the user equipment mayinclude some form of group bookmark list. The main purpose of the groupbookmark list is to allow the user to browse locally the list of groupsand easily attach to groups. Please note that there is no need for thegroup bookmark list in the user equipment to be complete and include allgroups available to the particular user. If a group is missing, the usercan access the group by giving its URL and then store it on the list.

[0186] From the user's point of view, the group bookmark list may beperceived as the traditional PMR group or channel selector. Other typesof users may perceive the list as a second phone book, an internetbookmark list or similar to TV channel settings. This set of models isenough to cover all likely users of the service.

[0187] There are a few options as to how the group bookmark list works,depending on the type of the intended market (PMR or consumer). For PMR,the user interface should resemble a traditional PMR group list, and afacility to remotely load new groups to the group list (from a systemmanager) will be needed. For consumer users, the user interface mightresemble more a bookmark list to which the user can add groups himself(e.g. bookmarking the group currently selected). A PMR user wouldnaturally also benefit from the facility of bookmarking the currentgroup. At this stage, we can assume that the deletion of groups from thelist is the user's responsibility. For PMR users, automatic bookmarkingcould be useful, i.e. that all new groups will automatically bebookmarked.

[0188] User Plane Operations

[0189] Signaling a PoC Group Speech Item

[0190] The user has to send all his user plane traffic to the U-UPF 20,and in case the traffic is destined to a group then the specific portnumber associated by the U-UPF 20 with the group is used for trafficidentification purposes.

[0191] One common PMR requirement is that only one active member at atime is allowed to speak in each group and that means that a userwilling to speak to a selected group has to get a speech item that ismanaged by the system. The speech items are granted and rejected by theG-UPF 21.

[0192] The straightforward way to support this functionality would be touse SIP signaling, but in order to avoid the delay introduced by theexplicit signaling transactions an alternative solution that uses thepayload type field and the payload itself of the RTP packet for implicitsignaling is here preferred.

[0193] Thus, in an embodiment of the invention, the system is not basedon request-grant type management of talk spurts such as is used inconventional PMR systems, i.e. TETRA. Rather, to provide fasteroperation, a user will start the transmitting talkspurts without atalkspurt grant from the system, but in case of clash of multipletalkers (or other problems) the right to transmit will be withdrawn.

[0194] With the implicit signaling approach mentioned above each usercan try to speak to the selected group whenever he wants. Referring toan example of FIG. 8, when the user of the MS pushes the PTT, the uplinkresources are reserved as described above, and the MS sends a leader RTPpacket to the G-UPF 21 via U-UPF 20. A speech item is available andgranted to the MS. At the same time, timers which are to be describedbelow are initialized. The leader RTP packet is forwarded to all thereceiving active members of the group via their respective U-UPFs, inorder to indicate the current speaker's identity to all the receivingactive members of the group each time an active member gets the speechitem and starts to talk to the group. This leader packet uses a specialpayload type for embedded control signaling as well as the RTP payload,to carry information about the sender's identity (mnemonic, number,etc.), and an SSRC value that will be used to recognize the followingRTP packets sent by the same speaker. Then the leader RTP packet isfollowed by the actual group's audio stream (VoIP RTP packets).

[0195] Normally, when the speech item is not available and the G-UPF 21does not grant the speech item to a user in response to receiving theleader RTP packet, the user notices that his voice is not forwarded whenhe receives another member's voice from the same group traffic. However,this would not be enough in case the user is simultaneously listening toanother group, so the G-UPF 21 of the group has to signal to the user'sU-UPF 20 (using embedded RTP signalling) that the user has not got thespeech item requested. The user's U-UPF 20 will then send this specialRTP packet forward to the user. This packet indicates to the userterminal that the speech item was not granted to him and allows the MSto switch on some hardware mechanism (such as visual or soundindication) to alert the user.

[0196] Since the speech item is managed by implicit signaling, there isno need for further specific explicit signaling during a groupcommunication.

[0197] Each G-CPF 23 generates unique SSRC values for the usersattaching to groups. During the group attachment the G-CPF 23 returnsthis SSRC value to the user and stores it in the G-CPF 23 and G-UPF 21.It should be noted that in this context the SSRC uniquely identifies theuser in the context of a group while a different SSRC associated by theU-UPF 20 for user's every one-to-one call indicates the caller in aone-to-one call, and the same value is used in that call for both thecaller and the called party.

[0198] The user's traffic forwarded by his U-UPF 20 is then identifiedby the IP address of the G-UPF 23 that is handling the group to whichthe traffic is destined, and the specific port number that the G-UPF 23has allocated for the traffic of that group.

[0199] Talkspurt Timers in Input Filtering

[0200] Traffic in a group, as seen by the users, consists of talkspurts(i.e. speech items) of more or less continuous speech coming from aspecific user. The U-UPFs 20 and G-UPFs 21, however, receive packets ofspeech, and multiple users may try to speak simultaneously in the samegroup. To ensure that speech from the current speaker in the group isnot interrupted or interfered with by packets from other users, theG-UPF 21 implements a talkspurt continuity timer for each active group.In addition to the timer, the identity of the currently talking user isstored.

[0201] In a typical talkspurt, while the user is pressing PTT, hisspeech codec is generating speech packets (frames) and these are beingsent at regular intervals. Of course, the packets will reach the G-UPF21 at somewhat more irregular intervals. Even when the user is notspeaking, the MS will be sending DTX packets (DiscontinuousTransmission). The timer is thus needed to keep a soft state betweenpackets. The timer is restarted for every incoming packet, and the timervalue should be enough to allow for the interval between packets, takinginto account the interval between packets sent (e.g. DTX frames) and thevariation of delay between the user and the G-UPF 21. The timer value isthus on the order of hundreds of milliseconds.

[0202] The idea is not to keep the turn reserved for the user if hereleases the PTT. Therefore, an embodiment of the invention uses atrailer packet to signal the end of the talkspurt, and this should thenbe considered equivalent to the expiry of the timer. The talkspurtcontinuity timer is implemented in G-UPF 21, because it is there thatdifferent talkers are contending for talking in the same group.

[0203] There is also a requirement to limit the maximum talkspurt time.From the user's point of view, no single user should be able to occupythe group unnecessarily long, preventing others from talking. Neithershould the group be blocked if the PTT of a user is unintentionallyjammed in the send position. The operator may want to restrict thetalkspurt duration for reasons of profiling the service and tariffing.Typical values for the talkspurt maximum timer would be 30 s, 60 s, evenmore. The timer is started at the first packet, when the user becomesthe current talker. At the expiry of the timer the talkspurt of thecurrent speaker will be stopped, even if there is no other speaker. Tobe able to talk again, he will need to release the PTT and push itagain. A special embedded RTP signaling packet is sent by the G-UPF tothe MS in order to stop the sending.

[0204] It is possible to implement the talkspurt maximum timer either inthe U-UPF 20 and in the G-UPF 21 or in both, but the result is not thesame in the two cases. A timer in the U-UPF 20 implements a userspecific maximum talkspurt duration, a timer in the G-UPF 21 implementsa group specific maximum talkspurt duration. Either of these can beuseful, even both. The architecture supports both

[0205]FIG. 9 is a flow diagram illustrating the talkspurt timer processin the G-UPF 21. The talkgroup continuity timer and the talkspurtmaximum timer are started when a speech item is granted to a user, step90. In step 91, it is checked whether a new packet has been receivedfrom the user. If not, it is checked whether the continuity timer hasexpired (step 92). If the continuity timer has expired, the speech itemis ended (step 97). If the continuity timer has not expired, it ischecked whether the maximum timer has expired (step 93). If the maximumtimer has expired, the speech item is ended (step 97). If the maximumtimer has not expired, the process returns to step 91. If a new packethas been received from the user in step 91, it is checked whether thereceived packet is a trailer packet sent by the user equipment inresponse to a release of the PTT (step 97). If a trailer packet has beenreceived, the speech item is ended (step 94). If the received packet isnot a trailer packet, it is checked whether the maximum timer hasexpired (step 95). If the maximum timer has expired, the speech item isended (step 97). If the maximum timer has not expired, the talkspurttimer is restarted (step 96) and the process returns to step 91.

[0206] A mechanism to interrupt a talkspurt may also be needed. Thiscould be when an authorized user needs to override an ongoing talkspurt.A G-CPF 23 may then be able to command the G-UPF 21 either 1) tointerrupt a talkspurt in a group, or 2) to set a user to have aninterrupting priority in a group. In case 1, any ongoing speech item inthe group shall be interrupted, as a consequence no-one has the speechitem and a command to stop transmitting (embedded RTP signaling packet)is sent to the MS of the interrupted talker. In case 2, a speech itemfrom the prioritized user will cause sending of a stop transmittingcommand to the previous talker, restarting of the talkspurt timers andgranting the speech item to the new talker. It is assumed thatinterrupting priority is only used temporarily, on demand or in specialcases. Therefore, the number of users with interrupting priority pergroup does not have to be large (assumption=1).

[0207] In the preferred embodiment of the invention, the command to stoptransmitting is achieved by the use of an RTP packet using a specialpayload. A parameter field in this packet may indicate the reason forcommand. These packets shall pass through any filtering processesunhindered. An MS receiving the command to stop transmitting immediatelystops transmitting voice packets and return to the receive state, as ifthe PTT were no longer pressed. The user will start to hear any incomingvoice traffic even if he holds the PTT pressed. To start transmittingagain, the user must first release the PTT and press it again. Thecommand to stop transmitting is either generated or routed via the U-UPF20.

[0208] Downstream Suppressing while Transmitting

[0209] The semi-duplex mode of operation makes it undesirable to loadthe downstream channels with voice packets while the user istransmitting. Therefore, in an embodiment of the invention, the U-UPF 20implements a suppression of downstream while transmitting in thefollowing way. Referring to FIG. 10, the upstream traffic from the userconcerning group calls is monitored for leader and voice packets (step101). An upstream talkspurt timer shall be used to provide a soft stateindicating that the user is transmitting a talkspurt in a group call.The mechanism used is the same as described above for maintainingtalkspurt continuity in a group. An upstream voice or leader packet willset the upstream talkspurt state ON (step 102), and start the upstreamtalkspurt timer (step 103). Expiry of the timer (step 104) will setupstream talkspurt state OFF (step 105). When the upstream talkspurtstate is ON, no downstream packets (except signaling) is sent to theuser. When the upstream talkspurt state is OFF, downstream packets aresent normally.

[0210] Audio Data Distribution in User Plane

[0211] In the user plane the audio data real-time distribution to/fromthe end users is handled, and the PoC Bridge 10 (the G-UPF 21 and theU-UPF 20) is the network element responsible for that. When multiplebridges/proxies are involved in the same PoC communication, their workis controlled and coordinated by the PoC CPS 11 (the G-CPF 23 or U-CPF22) that is handling the corresponding SIP session.

[0212] It is an object of the invention that the PoC approach isscalable to millions of users and at least hundreds of thousands ofgroups. To provide a scalable PMR solution a specific addressing modelhas been planned. The principal aim of this model is to implement thecomplex mapping between the bridges, the users and their traffics usingthe strictly needed amount of IP addresses and port numbers andpreferring static allocations (where possible) in order to reduce theamount of information to be exchanged between the network entities.

[0213] The IP/UDP/RTP protocol stack is commonly used in the VoIP worldfor real-time audio data transmission, and thus it is selected for theuser plane in the preferred embodiment of the invention as well.

[0214] In particular it is assumed that at least in the users' terminalsthe IPv6 is implemented, while in some core network entities it could berequired to support the IPv4 also (dual IPv6/v4 stack) in order toassure the interoperability with eventual subnetworks still using it.

[0215] The Real-time Transport Protocol (RTP) developed by the IETF tosupport the transport of real-time streams for audio communications overpacket networks is used on top of the UDP in order to avoid the delaysintroduced by more reliable transport protocols (not required in thiscontext), such as the TCP. With the RTP and latency buffering at thereceiving endpoint, the timing (jitter problem), packet ordering,synchronization of multiple streams, duplicate packet elimination andcontinuity of the streams can be handled.

[0216] When a user speaks to a group, the user's MS sends the audiopackets to his U-UPF 20 which after the input checking forwards it tothe group's G-UPF 21. The traffic forwarded by the U-UPF 20 is uniquelyidentified by the IP address of the G-UPF 21 and the port number theG-UPF 21 has associated with the group, while the traffic between theuser and his U-UPF 20 is identified by the IP address of the U-UPF 20and the port number the U-UPF 20 has associated with the group, so theMS can use the same socket to send and receive traffics from any groups(port number “200” is used in the following examples).

[0217] When a user becomes an active member of a group he gets from hisU-CPF the port number assigned by his U-UPF to the group's traffic. Andat the same time the U-CPF 22 and the G-CPF 23 set the proper mappingsbetween the user's U-UPF 20 and the group's G-UPF 21. More specifically,the U-UPF 20 gets the port number that the G-UPF 21 has assigned to thegroup's traffic.

[0218] A U-UPF 20 identifies incoming one-to-one traffic by the specificport number it has allocated for all one-to-one communications and theSSRC value assigned by the U-UPF 20 of the caller to the one-to-one callduring its establishment. In order to avoid the negotiation of dynamicport numbers between the MSs and the U-UPFs 20, a static port numbershall be used in all the MSs and U-UPFs (“102” in the followingexamples).

[0219] With the “split bridge” model described above it may happen thatin the downlink a G-UPF 21 has to forward the incoming group traffic toseparated U-UPFs 20. For that kind of communications, as well as forcommunications from a U-UPF 20 to G-UPF 21, port numbers assigned foreach group are used.

[0220] In order to better describe how group calls are managed on theuser plane, an example will now be illustrated. The current group'sspeaker sends his audio packet to his U-UPF 20 that checks the packetand forwards it to the group's G-UPF 21. If the traffic passes the inputfiltering in the G-UPF 21, then it is individually delivered to thescanning processes of the active members directly handled by the localU-UPF (located in the G-UPF physical entities). At the same time thetraffic is also forwarded to the other U-UPFs involved, which will thenserve their own active members.

[0221] An example of a group communication with two bridges involved isillustrated in FIG. 11. The G-UPF 1 and the associated U-UPF 1 have theIP address 1.0.0.1. The G-UPF 1 and the U-UPF 1 have ports 500 and 502,and 102, 700 and 702. The G-UPF 2 and the U-UPF 2 have ports 500, and102, 600 and 602. In both bridges port 102 is allocated for the purposesdescribed above. The other ports are allocated to groups G1, G2 and G3,as can be seen in FIG. 11. The Forward operation in some of the portsmeans that a packet received to the respective port must be forwarded toan IP address and the port indicated. Mobile stations MS1-MS4 have theIP addresses as well as the one-to-one ports and the group traffic portsshown in FIG. 11. The U-UPF 1 has been assigned to the Mobile stationsMS1 and MS4. The U-UPF 2 has been assigned to the Mobile stations MS2and MS3. The MS1 belongs to groups G1 and G2, the MS2 belongs to groupG1, the MS3 belongs to groups G1 and G3, and the MS4 belongs to groupsG1 and G2.

[0222] Let us now assume that the MS1 sends an audio packet 1 with adestination IP address 1.0.0.1 and a destination port 700. Consequently,the audio packet 1 is routed to the port 700 in the U-UPF 1. The port700 has been allocated to the group G1, and therefore the U-UPF 1multiplies the packet to all its users belonging to the group G1. Inthis case the audio packet 4 is sent to the MS4. The port 700 in theU-UPF 1 has also a Forward function to the IP address 1.0.0.2 and theport 500. Therefore, the U-UPF 1 sends a replica of the audio packet 1,i.e. the audio packet 2, to this destination. As a consequence, theaudio packet 2 is routed to the port 500 in the G-UPF 2. The port 500has been allocated to group G1, and therefore the U-UPF 2 multiplies thepacket to all its users belonging to the group G1. In this case theaudio packets are sent to the mobile stations MS2 and MS3.

[0223] The port 500 in the G-UPF 2 has also a Forward function to the IPaddress 1.0.0.1 and the port 700. Therefore, the G-UPF 2 sends a replicaof the audio packet 2 to this destination. As a consequence, the audiopacket is routed to the port 700 in the U-UPF 1.

[0224] Multi-Unicast

[0225] As described above, one aspect of the invention is that groupcommunication in a mobile radio system is implemented using a groupserver, which receives voice packets addressed to a group and eventuallyforwards (via U-UPF) these voice packets individually to each groupmember. The group server provides a number of groups (G1 . . . Gn). Agroup member sends voice packets to the group server (via U-UPF); eachpacket is addressed to the group server but it also carries the identityof the group (G1 . . . Gn). The group server holds a table, for eachgroup, containing U-UPF addresses of group members and U-UPF holds atable, containing individual addresses of users.

[0226] In the example illustrated in FIG. 12, the source MS sends justone stream in the uplink, but then the PoC Bridge multiplies it foreight different recipients. On top of each link the number of streamstransported in the uplink and in the downlink are respectivelyindicated, and each receiving MS is labeled with the number of hopsneeded by the packets to reach the respective MS. The reference symbol Rrepresents any routing node in the system.

[0227] This concept is called multi-unicast herein. This concept is anon-traditional method for implementing group communication in a mobileradio network, and it greatly reduces the complexity of implementationof group communication services.

[0228] The core issue of group communication is how to deliver thecommunication to group members in an efficient manner in a mobile radiosystem. These systems may range in size from very small (from one basestation) to nation-wide (thousands of base stations). Likewise thegroups may be of different, even varying sizes. Even more difficult, thegeographical distribution of a group can be anything from very local tonation-wide, and vary according to the circumstances. In other words,the problem in hand is how to deliver the group traffic to each groupmember reliably regardless of the location of the member and thedistribution of the members.

[0229] Traditional radio systems were small and groups usually local.Therefore the obvious solution was to use one transmission per basestation for each group active in a specific area. The transmission wasidentified by a group address (multicast). The prior art approachinvolves many problems in a large communications system. Firstly, when agroup call is made, the system needs to know which base stations to usefor the call. Thus the system needs to implement a separate mobilitymanagement subsystem to keep track of the location of group members foreach group. This causes a significant increase of the complexity of thesystem and can become the primary factor in the total processing load.Secondly, in order to receive group traffic, a mobile station has tohold the proper group addresses. Therefore, for everything to workproperly, group membership must be known beforehand to both the mobilestation and all the relevant system elements. This requires adistributed data management subsystem which has to operate over anunreliable and very low bandwidth radio channel.

[0230] These problems characterize the current state of the technology.The prior art systems circumvented the problem by not trying to optimizethe use of base stations at all. Traffic of a group was radiated on afixed, predefined set of base stations, thus relieving the need formobility management for groups. This meant also that the system did nothave to know the group members, and the group addresses were programmedinto the mobile stations.

[0231] By means of the multi-unicast concept according to the invention,group communication in a large mobile system can be implemented reliablywithout adding large subsystems, which cause huge processing load andare prone to errors during operation—giving the users an experience ofunreliable service. Because group traffic be delivered to recipientsusing the individual addressing and the basic mobility management of thesystem, group traffic becomes as reliable as individual traffic.

[0232] It can be argued that using individual delivery is more resourceconsuming than multicast delivery. This certainly was true intraditional PMR systems which were based on a large cell size; thereforea significant number of group members could be located within the rangeof a single base station. In modern cellular networks the use of largecells is inefficient from point of view of the frequency utilization,smaller and smaller cells are being deployed and therefore theprobability of group members being located in the same cell isdecreasing.

[0233] It should be noted that the basic architecture of the inventioncan also use some multicasting mechanism for audio data distribution,but that would require muticasting functionalities at the RAN with theabove problems. Anyway, in this case it may be still reasonable tosupport both unicast (multi-unicast) and multicast distributiontechniques in order to get benefit of unicasting where it is moreefficient, for example when few members of a group have to be served ata site, or where multicasting is eventually not supported.

[0234] Scanning Filtering

[0235] In the current PMR system all traffic addressed to a user isdelivered to his terminal which locally performs the filtering functionto play out the single traffic that the user wants to listen to. Thistask has to be done according to the user's scanning settings and has tosupport the eventual overriding of the incoming traffic from higherpriority groups or emergency calls.

[0236] In order to avoid the waste of bandwidth in the downlink for thetransmission of traffic that will not be played out in the terminal, thefiltering function obviously has to be implemented beforehand in thenetwork, and this is one of the motivations for introducing the PoCBridge 10 into the network architecture according to the preferredembodiment.

[0237] The role of the bridge 10 in this context could be seen as twoserial processes, namely group and user specific processes, asillustrated in FIG. 13. In the group specific process the G-UPF 21 hasto multiply an incoming traffic in several packet streams which have tobe forwarded to all the active members of the group or, in the preferredembodiment of the invention, to U-UPFs having active members in thegroups to which these traffic streams are destined. In the user specificprocess, the U-UPF 20 has to decide which one of the several possibletraffic streams addressed to a user actually needs to be forwarded tohim. U-UPF also multiplies the stream for every user who receive thegroup traffic according to his current scanning settings (in the casethe U-UPF serves more than one user). Sent traffic is normally thetraffic from the currently listened group, but occasionally could be anoverriding traffic stream.

[0238] In order to ensure conversation continuity (i.e. to ensure that alistener receives a coherent series of transmissions), a specific timeris provided in the U-UPF 20. The function of this timer is to keep theuser receiving consecutive talkspurts in the same group (or individualcall) unless there is a pause longer than a certain timeout in theconversation. Here we are talking about typical values between 2 and 15seconds.

[0239] In principle this means that the scanning process shall lock tothe received group after each packet, for the duration of this timer.Timer is located in U-UPF and timer values are preferably groupspecific. It is also advisable to use a different timeout for group andindividual traffic. However, when higher priority traffic than thecurrently listened stream addressed to user arrives in U-UPF, higherpriority stream overrides lower priority traffic immediately and theconversation continuity timer has no effect.

[0240] An example of the implementation of the scanning filteringprocess is illustrated in FIG. 14. In step 141, the process chooses oneof multiple (i.e. two or more) voice packet traffic streams arriving tothe U-UPF 20 from the G-UPF(s) 21 (group communications) or from anotheruser or U-UPF (one-to-one communication), and forwards the chosentraffic stream to the user. Other arriving traffic streams arediscarded, i.e. not forwarded to the user. When the choice is made, atimer is set to a predetermined value “Pause period”, i.e. a maximumperiod of time between two consecutive voice packets in the chosentraffic stream (step 142). In step 143, the process checks whether a newRTP packet has been received. If a new RTP packet has arrived, it isfirst checked in step 145 whether that packet belongs to a higherpriority stream than the previous packet. If this packet does not havehigher priority than the previous one according to user's scanningsettings, the process moves on to step 146 where the timer is reset andthe packet is sent to the user. After that the process returns to step143. If in step 145 it is noticed that the new packet belongs to astream having higher priority than the previous one, the new packet issent to the user and the timer is reset (step 147). After that theprocess returns to step 143. If no new packet is received, it is checkedwhether the timer has expired (step 144). If the timer has not expired,the process returns to step 143. If the timer has expired, the processdeems the selected traffic stream to be interrupted, and returns to step141 to select a new traffic stream.

[0241] One-to-One Call Management

[0242] An example of one-to-one call management is now described withreference to FIG. 15. A static port number is allocated into each U-UPF20 for one-to-one traffic (such as the port 102 in FIG. 11).

[0243] If a user wants to establish a one-to-one communication, hepushes the PTT in his terminal MS1. The MS1 has just to send a leaderpacket containing his identity information (number or name) to hisU-UPF1 using the specific “one-to-one” port number 102. This specialleader packet is identified as such by the use of a specific RTP payloaddesignated for this purpose. In addition to the identity of the calledparty (MS2), the leader packet may contain other relevant information.

[0244] Firstly, the caller's U-UPF1 assigns an SSRC value to be used byboth participants in this one-to-one call. For reaching the called partyand for performing the necessary rights checks, user's U-UPF1 is nowcontacted. It in turn contacts the SGMF for obtaining the called party'sU-CPF2 address, for rights checking and for defining the correct formfor caller's name representation. The information itself is contained inthe PoC database or directory (PoC main information repository), fromwhere SGMF gets the necessary information. The information is returnedto caller's U-CPF1.

[0245] Now, U-CPF2 of the called party is contacted using SIP inviterequest. U-CPF2 of the called party sends a message to U-UPF2 of thecalled party, which in turn sends a leader packet to MS2 for checkingits ability to receive one-to-one call. Also an acknowledgement is sentto U-CPF2 of the called party, which in turn returns a SIP OK message toU-CPF1 of the caller. U-CPF1 of the caller sends a message to U-UPF1 ofthe caller, and U-UPF1 acknowledges the message. Finally, SIPacknowledgement is returned to U-CPF2 of the called party and thenetwork has successfully setup the call.

[0246] After a positive (embedded RTP signaling) acknowledgement isreceived from U-UPF of the called party, it is forwarded to MS1 whichcan now begin sending voice RTP packets.

[0247] At this stage the calling party is talking and the terminal MS1sending RTP packets containing voice to his U-UPF 20, which based on theSSRC field in the packet will send the voice RTP packets to the calledparty's (MS2) U-UPF 20. Afterwards the called party's U-UPF willeventually deliver them (depending on the scanning process results) tothe called party's terminal.

[0248] The called party ends the communication by releasing the PTT, inwhich case MS1 sends a trailer packet in order to indicate the stop ofthe communication to the U-UPF. It is also possible to employ acontinuity monitoring as described with respect to FIG. 9.

[0249] Let us now consider examples of some special signaling caseswhich may occur in one-to-one communication.

[0250] The called party may simultaneously receive traffic in a group.Because of the limited bandwidth of the downlink, it is not advisable toforward multiple voice streams to the same mobile station MS (unless itis known that there is enough bandwidth to support reception of multiplestreams). Therefore, in one-to-one traffic in the downlink is routedthrough the same scanning process in the called user's U-UPF 20, asapplies to the group traffic. This ensures that each MS is only beingsent one voice stream at a time.

[0251] Similarly, the same called party may receive more than oneone-to-one call to the same called party at the same time. Therefore,the called party's U-UPF shall detect if there is a one-to-one voicetransfer ongoing to an MS, and prevent any simultaneous one-to-onestreams to the same MS. This is preferably handled with the same processthat prevents multiple talkers in a group (the incoming traffic in theone-to-one port is filtered according to the recognized SSRCs).

[0252] Failing of the one-to-one call setup can depend on many differentreasons, in which cases the caller's U-UPF receives a negative (embeddedRTP signaling) acknowledgement from called party's U-UPF, which isforwarded to MS1. An example of this is that the the scanning process ofMS2 is forwarding higher priority traffic. Another examples of thisinclude that 1) the called party is unknown, 2) the called party is notcurrently logged on to the PoC service, 3) call rights check indicatesthat one-to-one calls between the parties are not allowed and 4) thecalled party is engaged in a circuit mode call. To ensure that thecommunicating parties experience a sense of mutual, two-waycommunication, U-UPF shall implement timer to ensure that a speech itemthat has been allowed to start (packets are being forwarded) is notinterrupted by any traffic (except when overridden by higher prioritytraffic). Additionally, caller's U-UPF implements timers for ensuringthat aa) conversation is not interrupted between short breaks (of theorder of some seconds) between the speech items, b) speech itemmanagement is performed (either one of the participants has talkspurtstate on) and c) maximum talkspurt time is observed (either one of thecall participants is prevented from talking for too long. It must benoted that when timer a) goes off, the one-to-one call is cleared in PoCnetwork.

[0253] Security

[0254] It is a requirement that the users should be able to rely, up toa reasonable level, on the identifications (group, talking party e.g.)provided by the system. The users should be able to rely, up to areasonable level, that the contents of the received data have not beentampered with. The reasonable level corresponds to what is provided bypublic, circuit switched telephone networks.

[0255] Two principal approaches have been identified to satisfy thisrequirement: 1) relying on the security provided by the RAN and thesecurity provided by the IP network between the RAN and the CPS 11 orthe bridge 10; and 2) using the Security Architecture for IP (IPSec)authentication between the user equipment (the MS) and the CPS 11 or thebridge 10.

[0256] Relying on the security of both the underlying RAN and the IPnetwork means specifically that 1) the CPS 11 and the bridge 10 checkthe identity of the transmitting user by looking at the source IPaddress; therefore the network prevents spoofing the source IP address;2) an MS checks the identity of the transmitting CPS 11 or the bridge10; therefore the network prevents spoofing the source IP address; 3)and the underlying network does not easily allow tampering with thecontents of the packets.

[0257] The majority of users do not require extreme security. Usually asatisfactory level is achieved by air interface encryption in the RANand preventing outside access to the traffic in the IP network. Ifnecessary, the security of the IP network can be improved by using theIPSec between the network elements (this applies both to the IP networkand to the PoC elements: the CPS 11 and the bridge 10).

[0258] As an option, the architecture according to the invention allowsusing the IPSec Authentication Headers (AH) between the MS and its U-UPF20. Each MS (or the user, if needed) has a public-private key pair;likewise the U-UPF 20 has a public-private key pair. Standard IPSecmechanisms can then be used to set up a security association between theMS and its U-UPF 20.

[0259] This arrangement allows the authentication of an MS (or a user)which is logging on to the PoC service. After the log-on, the IPSecauthentication headers must be used in all packets from the MS to theU-UPF 20. In this way the origin and integrity of the packets arrivingat the proxy 20 (or CPS in case of control) can be verified. Similarly,authentication headers can be used in all packets from the proxy to theMS, which allows the MS to verify the origin and the integrity of thepackets. In this manner, the security becomes a matter of trust betweenthe MS and the proxy.

[0260] In other words, each MS subscribing to the PoC service has atwo-way Security Association (SA) with its U-UPF 20. A complete workingsetup will require in addition: 1) A means to set up and manage thesecurity associations (Internet Key Exchange, IKE); 2) A means to verifythe public keys with a trusted source; and 3) A means to generate anddistribute the public and private keys (Internet Security Associationand Key Management Protocol, ISAKMP).

[0261] Even if this mechanism may look complicated, it uses standard andreadily available solutions. If the IPSec authentication is taken intouse later, it can be taken into use gradually by installing it in oneU-UPF 20, which will be used to serve users with IPSec only. In otherwords, it is possible to support both authenticating andnon-authenticating MSs and proxies.

[0262] Encryption

[0263] It is a requirement that specific users be able to use end-to-endencryption. As a simpler alternative, two-leg end-to-bridge-to-endencryption should be considered, because this greatly simplifies keymanagement.

[0264] For users with higher encryption requirements than what isprovided by air interface encryption of the underlying network, IPSecEncapsulating Security Payload (ESP) can be used to provideconfidentiality (encryption) between the MS and the bridge. This, ofcourse, requires that the IPSec is in use.

[0265] For instance, MS1 will encrypt the payload of the voice packetsfor sending to the proxy. The proxy will decrypt the packets, and thenencrypt them again for forwarding to MS2. This provides almost the samelevel of security to the users as does end-to-end encryption, withoutany need for the communicating parties to share keys. Therefore, theusual key management problems associated with end-to-end encryption donot exist in this model.

[0266] In the proposed model, the PoC elements (CPS, Bridge) are theonly security critical components in the network. Therefore, for userswith very high security requirements, it might be feasible to installseparate user proxies and bridges on secure premises under the controlof the user group.

[0267] The description only illustrates preferred embodiments of theinvention. The invention is not, however, limited to these examples, butit may vary within the scope and spirit of the appended claims.

1. A method for a packet mode group voice communication in acommunications system, comprising the steps of providing a groupcommunication service entity on top of the said communications system,providing said group communication service entity with individualaddresses of group members in at least one group communication group,sending voice packets from one of said group members to said groupcommunication service entity, each voice packet being addressed to saidat least one group, forwarding said voice packets individually to eachreceiving one of said group members on the basis of said individualaddresses.
 2. A method according to claim 1, wherein said step offorwarding comprises a step of forwarding said voice packetsindividually by user communications functions provided on top of thesaid mobile communications system, said user communications functionsmanaging user-specific voice packet streams to and from users.
 3. Amethod for packet mode group voice communication in a communicationssystem, comprising the steps of providing group communication serviceentity with individual addresses of group members of a groupcommunication group, creating an individual logical connection from eachgroup member to said group communication service entity by means ofoutband signaling, starting a speech item in said group by sending aleader packet embedded in a user traffic stream from one of said groupmembers to said group communication service entity over said individuallogical connection, each leader packet containing the identifier of therespective group member, said group communication service entity eitheri) rejecting said started speech item, or ii) granting the startedspeech item to said one group member and forwarding said leader packetand subsequent voice packets in said user traffic stream individually toeach receiving one of said group members in said group on the basis ofsaid individual addresses.
 4. A method according to claim 3, comprisingthe further steps of allocating an uplink bearer for said one groupmember in an air interface of said communications system prior to saidone group member sends said leader packet and prior to said granting ofsaid speech item, and allocating a downlink bearer in an air interfacefor each receiving group member in response to receiving a leader packetforwarded by said group communication service entity and addressed tosaid respective group member, said leader packet being embedded to auser traffic stream.
 5. A method of managing speech items in acommunications system having a packet mode group voice communicationfeature, comprising the steps of granting a speech item to one groupmember of said group communication group, setting a first timer tomeasure a predetermined idle period in response to said granting,resetting said first timer each time a voice packet is received fromsaid one of said group members, ending said granted speech item if saidfirst timer expires indicating that said predetermined idle period haselapsed from said granting or from last reception of a voice packet fromsaid one group member.
 6. A method according to claim 5, comprising thefurther step of ending said granted speech item if a maximum allowedperiod of time has elapsed from the granting.
 7. A method according toclaim 5, comprising the further steps of said one group member sends atrailer packet having a predetermined payload in order to indicate theend of sending, ending said speech item in response to receiving saidtrailer packet.
 8. A method of managing traffic streams in acommunications system having a packet mode group voice communicationfeature, comprising the steps of providing a user-specificcommunications function for managing traffic streams addressed to a userwho is active in at least one group communication group or in aone-to-one communication, receiving a first voice packet stream relatedto a first group communication group or a first one-to-one communicationand addressed to a user who is active at least in said first groupcommunication group or in said first one-to-one communication,forwarding said first voice packet stream to said respective user,monitoring continuity of said first voice packet stream, receiving atleast one further voice packet stream related to at least one furthergroup or one-to-one communication, forwarding no one of said at leastone further voice packet streams to said user if said first voice packetdata stream is continuous, forwarding one of said at least one furthervoice packet streams to said user if said first voice traffic stream hasbeen discontinued for a predetermined period of time.
 9. A methodaccording to claim 8, wherein said step of monitoring comprises thefurther steps of setting a timer to measure said predetermined period oftime when a first packet of said first voice packet stream is forwardedto said user, resetting said timer each time a new packet of said firstvoice packet stream is forwarded to said user, determining said firstvoice packet stream to be discontinued if said timer expires.
 10. Amethod according to claim 8 or 9, said method comprising a further stepof interrupting said first voice packet stream immediately when a voicepacket stream having higher priority is received.
 11. A server systemfor providing a packet mode group communication service for acommunications system, said server system comprising a group serverprovided on top of said communications system, said group server furthercomprising means for storing individual addresses of group members in atleast one group communication group, means for receiving voice packetsfrom said group members, each received voice packet containinginformation identifying the communication group which the respectivepacket is addressed to, means for granting a speech item to one groupmember per a communication group in turn, means for unicasting eachvoice packet received from said group member having a speech item in agroup communication group separately to each receiving member in saidrespective group communication group on the basis of said individualaddresses.
 12. A server system according to claim 11, wherein saidinformation identifying the communication group identify a port assignedto said group in said group server.
 13. A server system according toclaim 11, further comprising a call processing server provided on top ofsaid mobile communications system, said call processing server beingresponsible for control plane management of the group communications insaid group server.
 14. A server system according to claim 11, whereinsaid means for granting a speech item further comprises a first timerresponsive to said granting to start measurement of a predetermined idleperiod from said granting, means for resetting said first timer eachtime a voice packet is received from said one group member having saidgranted speech item, means for ending said granted speech item if saidfirst timer expires indicating that said predetermined idle period haselapsed from said granting or from the last reception of a voice packetfrom said one group member.
 15. A server system according to claim 13,said system further comprising means for establishing an individuallogical connection from each group member to said group server by meansof outband signaling carried out between said call processing server andeach group member.
 16. A server system according to any one of claims 11to 15, wherein said means for granting a speech item further comprisesmeans for receiving a leader packet starting a speech item in said groupfrom one of said group members to said group server, said leader packetcontaining identifier of the respective group member and being embeddedin a user traffic stream, means for either i) rejecting said startedspeech item, or ii) granting said started speech item to said one groupmember and forwarding said leader packet and subsequent voice packets ofsaid user traffic stream individually to each receiving one of saidother members in said group on the basis of said individual addresses.17. A server system according to any one of claims 11 to 16, whereinsaid packets are real time transport (RTP) packets.
 18. A server systemaccording to claim 11, further comprising a group management serverproviding a user interface for a remote creation and management of groupcommunications group in said server system.
 19. A server systemaccording to claim 18, wherein said user interface is based on one ofthe World Wide Web (WWW) and Wireless Application Protocol (WAP)technologies.
 20. A server system according to claim 11, wherein saidgroup server is interconnected to said communications network by anInternet Protocol (IP) based network.
 21. A server system for providinga packet mode group communication service for a communications system,said server system comprising a group server provided on top of saidcommunications system, said group server further comprising means foridentifying and authenticating a source of group communication, meansfor controlling that only one group member in a group talks at a time,means for checking active group members in a group to which voicepackets from a currently talking group member are destined to and meansfor generating from an incoming voice packet an outgoing packet to beforwarded separately to each of said active group members, and means forselecting from possible multiple incoming traffic streams destined toone group member the one which is to be forwarded to said one groupmember.
 22. A server system for providing a packet mode groupcommunication service for a communications system, said server systemcomprising at least one first group communication network entityproviding group specific communications functions, said first groupcommunication network entity further comprising a data memory storingindividual addresses of group members in at least one groupcommunication group, means for receiving voice packets from said groupmembers, each received voice packet containing information identifyingthe communication group which the respective packet is addressed to,means for granting a speech item to one group member per communicationgroup in turn, means for unicasting each voice packet received from saidgroup member having a speech item in a group communication groupseparately to each receiving member in said respective groupcommunication on the basis of said individual addresses, at least onesecond user communication network entity providing user-specificcommunications functions for at least one user, whereby any grouprelated communication from a user managed by said second user networkentity being routed first to said second user network entity and thenforwarded to an appropriate first group network entity, and any unicastvoice packet from said at least one first group network entiy beingrouted first to said second user network entity prior to sending thevoice packet to the respective user.
 23. A server system according toclaim 22, wherein said means for unicasting in said first group networkentity comprises means for unicasting each voice packet received fromsaid group member having a speech item in a group communication groupseparately to each second user network entity serving at least one userwhich is a group member in said respective group, each second usernetwork entity being arranged to multiply the voice packet for eachgroup member and send the voice packet to the respective users.
 24. Aserver system according to claim 22 or 23, wherein said informationidentifying the communication group identify a port assigned to saidgroup in said group server.
 25. A server system for providing a packetmode group communication service for a communications system, saidserver system comprising wherein at least one group communicationnetwork entity providing group specific communications functions, saidgroup network entity further comprising means for controlling that onlyone group member in a group talks at a time, means for checking activegroup members in a group to which voice packets from a currently talkinggroup member is destined to and for generating from an incoming voicepacket an outgoing packet to be forwarded separately to user serverhaving serving at least one active member in said group gs, a usercommunication network entity providing user-specific communicationsfunctions on a user plane for at least user, said user network entityfurther comprising means for identifying and authenticating a source ofgroup communication, means for selecting from possible multiple incomingtraffic streams destined to one group member the one which is to beforwarded to said one group member.
 26. A server system according toclaim 25, said system further comprising a group call processing entityresponsible for control plane management of the group communications insaid group network entity, and a user call processing entity responsiblefor control plane management of the communications in said user networkentity.
 27. A server system according to claim 25 or 26, wherein saidmeans for controlling speech items further comprises a first timer meansresponsive to a grant of a speech item for starting to measure apredetermined idle period from said granting, means for resetting saidfirst timer each time a voice packet is received from said one groupmember having said granted speech item, means for ending said grantedspeech item, if said first timer expires indicating that saidpredetermined idle period has elapsed from said granting or from lastreception of a voice packet from said one group member.
 28. A serversystem according to any one of claims 25 to 27, said system furthercomprising means for establishing an individual logical connectionbetween each group member and said user network entity by means ofoutband signaling carried out between said user call processing entityand each group member.
 29. A server system according to any one ofclaims 25 to 28, wherein said means for granting speech items comprisesmeans for receiving a request for a speech item in said group from oneof said group members to said group communication network entity in formof a leader packet embedded in the user traffic stream and containingidentifier of the respective group member, means for either i) rejectingsaid request for a speech item, or ii) granting the speech item to saidone group member and forwarding said leader packet and subsequent voicepackets of said user traffic stream individually to each receiving oneof said other members in said group.
 30. A server system according toany one of claims 25 to 29, wherein said packets are real time transportprotocol (RTP) packets.
 31. A server system according to any one ofclaims 25 to 30, said server system further comprising group managemententity providing a user interface for a creation and management of groupcommunications group in said server system.
 32. A server systemaccording to claim 31, wherein said user interface is based on one ofthe World Wide Web (WWW) and Wireless Application Protocol (WAP)technologies.
 33. A server system according to any one of claims 25 to32, wherein said group communication network entity is interconnected tosaid communications network by an Internet Protocol (IP) based network.34. A server system for providing a packet mode group communicationservice for a communications system, said server system comprising atleast one group communication network entity providing group specificcommunications functions in a user plane, said group network entityfurther comprising means for storing individual addresses of groupmembers in at least one group communication group, means for receivingvoice packets from said group members, each received voice packetcontaining information identifying the communication group which therespective packet is addressed to, means for granting a speech item toone group member per communication group in turn, means for unicastingeach voice packet received from said group member having a speech itemin a group communication group separately to each receiving member insaid respective group communication on the basis of said individualaddresses, a user communication network entity providing user-specificcommunications functions on a user plane for at least one user, wherebyany group related communication from a user managed by said user networkentity being routed first to said user network entity and then forwardedto an appropriate group network entity, and any unicast voice packetfrom said at least one group network entity being routed first to saiduser network entity prior to sending the voice packet to the respectiveuser, a group call processing entity responsible for control planemanagement of the group communications in said group network entity, anda user call processing entity responsible for control plane managementof the communications in said user network entity.
 35. A network unitfor managing speech items in a communications system having a packetmode group voice communication feature, comprising means for storingindividual addresses of group members in at least one groupcommunication group, means for receiving voice packets from said groupmembers, each received voice packet containing information identifyingthe communication group which the respective packet is addressed to,means for granting a speech item to one group member per a communicationgroup in turn, means for unicasting each voice packet received from saidgroup member having a speech item in a group communication groupseparately to each receiving member in said respective groupcommunication group on the basis of said individual addresses.
 36. Anetwork unit according to claim 35, wherein said means for receiving arearranged to receive voice packets from a group member via a usercommunication entity providing user-specific services for that groupmember, means for unicasting are arranged to unicast voice packetsseparately to each receiving member via a user communication entityproviding user-specific services for that group member.
 37. A networkunit according to claim 35 or 36, comprising means for granting a speechitem to one group member in group communication group at time, a firsttimer means responsive to said granting for starting to measure apredetermined idle period from said granting, means for resetting saidfirst timer each time a voice packet is received from said one of saidgroup members, means for ending said granted speech item, if said firsttimer expires indicating that said predetermined idle period has elapsedfrom said granting or from last reception of a voice packet from saidone group member.
 38. A network unit according to any one of claims 35to 37, wherein said network unit is on top of said communication system.39. A network unit according to any one of claims 35 to 38, wherein saidnetwork unit is connected to said communications network by an InternetProtocol (IP) based network
 40. A network unit for managing trafficstreams addressed to a user who is active in at least one groupcommunication group or in one-to-one communication, comprising means forselecting for unicast to a user a first voice packet stream related to afirst group or one-to-one communication addressed to said user, meansformonitoring continuity of said selected first voice packet stream,means for discarding any other received voice packet stream related toat least one further group or one-to-one communication, if saidcurrently selected voice packet stream is continuous, and means forselecting and unicasting another received voice packet stream to saiduser if said initially selected and unicasted first voice traffic streamhas been discontinued for a predetermined period of time.
 41. A networkunit according to claim 40, wherein said monitoring means furthercomprises a timer which is set to measure said predetermined period oftime when a first packet of said selected voice packet stream isforwarded to said user, means for resetting said timer each time a newpacket of said selected voice packet stream is forwarded to said user,means for determining said selected voice packet stream to bediscontinued if said timer expires.
 42. A network unit according toclaim 41, further comprising interrupting said first voice packet streamimmediately when a voice packet stream having higher priority isreceived.
 43. A method for establishing a one-to-one voice communicationin a communications system, comprising the steps of providing acommunication server on top of the said mobile communications system,creating an individual logical connection between said communicationserver and each user having an active communication service in saidcommunication server, starting a communication by sending a leaderpacket embedded in a traffic stream from a user to said communicationserver over respective said individual logical connection, each leaderpacket containing identifier of said sending user and a receiving user,said communication server either i) rejects said started speech item, orii) grants the started speech item to said sending user and forwardssaid leader packet and subsequent voice packets of said user trafficstream to said receiving user on the basis of said received identifierof said receiving user.
 44. A method according to claim 43, wherein saidstep of forwarding comprises the further steps of inquiring an IPaddress of said receiving user from a communication control server onthe basis of said received identity of said receiving user, forwardingsaid leader packet and subsequent voice packets to said IP address ofsaid receiving user.
 45. A method according to claim 43, wherein saidsending user sends the leader packet and the subsequent packets to aspecific port assigned for one-to-one communication in saidcommunication server.
 46. A subscriber equipment for communicationssystem having a packet mode group voice communication service, saidsubscriber equipment comprising mechanisms for packet data communicationover a communications system, a group communication application on topof said mechanisms, said application having first means for establishinga logical packet connection to a group communication server, saidapplication having second means for sending and receiving voice packetsto and from said group communications server.
 47. A subscriber equipmentaccording to claim 46, further comprising said application is a Voiceover IP (VoIP) application.
 48. A subscriber equipment according toclaim 47 or 48, further comprising a push-to-talk means, means for,reactive to activation of said push-to-talk means by a user, sending aleader packet followed by voice packets in a user traffic stream to saidgroup communication server over said logical connection and therebystarting a speech item.
 49. A subscriber equipment for communicationssystem having a packet mode group voice communication service, saidsubscriber equipment comprising a push-to-talk means, means, responsiveto activation of said push-to-talk means by a user, for sending a leaderpacket followed by voice packets in a user traffic stream to said groupcommunication service and thereby starting a speech item.
 50. Asubscriber equipment according to any one of claims 46 to 49, furthercomprising means, responsive to receiving an indication that a speechitem is not granted to the user from said group communication serviceafter sending said leader packet, stops sending further packets andstops the speech item although the push-to-talk means is stillactivated.
 51. A subscriber equipment according to any one of claims 46to 50, further comprising said third means, which, reactive todeactivation of said push-to-talk means by the user, stops the speechitem and stops sending further voice packets.
 52. A subscriber equipmentaccording to any one of claims 46 to 51, comprising means, reactive todeactivation of said push-to-talk means by the user, for sending atrailer packet embedded in a user traffic stream to said groupcommunication service and thereby stopping the speech item.
 53. Asubscriber equipment according to claim 50, wherein said indication is areception of a voice or leader packet originating from another user in agroup communication group after sending said leader packet.
 54. Asubscriber equipment according to claim 50, wherein said indication isthe reception of a voice packet having predetermined payload type aftersending said leader packet.
 55. A subscriber equipment according toclaim 50, comprising means, responsive to the reception of saidindication, for alerting the user of the fact the speech item was notgranted.
 56. A subscriber equipment according to any one of claims 46 to55, further comprising means for giving an audible indication to theuser to start speaking after the activation of said push-to-talk switch.57. A subscriber equipment according to claim 56, wherein saidindication means comprises a timer enabling said audible indicationafter a predetermined period of time has expired from said activation ofsaid push-to-talk switch.
 58. A subscriber equipment according to claim56, wherein said indication means gives said audible indication afterone of the connection setup phases has been reached; 1) after an uplinkbearer has been allocated, 2) after said leader packet has been sent, 3)after said group communication service has processed said leader packetand granted a speech item, 4) after a receiving party has acknowledgedsaid leader packet.
 59. A method for providing a packet mode groupcommunication service for a communications system, comprising storingindividual addresses of group members in at least one groupcommunication group, managing said group communication groups using acontrol plane signalling, group member requests speech item usinguser-plane signalling embedded in a user traffic stream, granting aspeech item to one group member per a communication group in turn basedon said embedded user plane signalling, receiving voice packets from agroup member having a speech item in a group communication group, eachreceived voice packet containing information identifying thecommunication group which the respective packet is addressed to,unicasting said embedded user-plane signalling and each voice packetreceived from a group member having a speech item separately to eachreceiving member in said respective group communication group on thebasis of said individual addresses.
 60. A system for providing a packetmode group communication service for a communications system, comprisingmeans for storing individual addresses of group members in at least onegroup communication group, means for managing said group communicationgroups using a control plane signalling, means for granting a speechitem to one group member per a communication group in turn based onspeech item requests speech sent by said group members using user-planesignalling embedded in a user traffic stream, means for receiving voicepackets from a group member having a speech item in a groupcommunication group, each received voice packet containing informationidentifying the communication group which the respective packet isaddressed to, means for unicasting said embedded user-plane signallingand each voice packet received from a group member having a speech itemseparately to each receiving member in said respective groupcommunication group on the basis of said individual addresses.
 61. Asystem according to claim 60, wherein the embedded speech item signalingcomprises a leader packet sent in a beginning of a user traffic streamcontaining user voice data packets, such as RTP packets, and wherein thesystem grants or rejects the speech item based on the leader packet. 62.A system according to claim 61, wherein the system, upon granting agroup communication service a speech item based on the leader packet,opens a speech item communication to receiving members of a group byforwarding a user traffic stream containing said leader packet andsubsequent voice packets to the receiving members.
 63. A systemaccording to claim 60, 61 or 62, wherein the embedded speech itemsignaling comprises a trailer packet sent at end of a user trafficstream containing user voice data packets, such as RTP packets, andwherein the system ends the speech item based on the trailer packet. 64.A system according to claim 60, 61, 62 or 63, wherein the systemforwards a trailer packet at end of a user traffic stream to receivinggroup members in order to end the speech item communication to thereceiving group members.
 65. A subscriber equipment for communicationssystem having a packet mode group voice communication service, saidsubscriber equipment comprising mechanisms for packet data communicationover a communications system, a group communication application on topof said mechanisms, said application having first means for establishinga logical packet connection to a group communication service by means ofa control plane signalling, said application having second means forsending and receiving voice packets to and from said groupcommunications server, said application having third means forrequesting a speech item by means of a user plane signalling.
 66. Asubscriber equipment according to claim 65, wherein the embedded speechitem signaling comprises a leader packet sent in a beginning of a usertraffic stream containing user voice data packets, such as RTP packets.67. A subscriber equipment according to claim 65 or 66, wherein theembedded speech item signaling comprises a trailer packet sent at end ofa user traffic stream containing user voice data packets, such as RTPpackets.